[SIPForum-discussion] G 723 Audio Codec Bit Rate

[sreekant nair]

Hi All, 

Based on the inputs received from multiple people, we went ahead with our design. However during one of the tests we saw the problem being discussed. 

The G.723.1 codec dynamically changed its bit rate on the fly between two successive RTP frames. I have captured the trace file and exported it into a text format for easy viewing for all. Please see the portion highlighted in the document. 

Any thoughts are greatly appreciated. 

Thanks

Sreekant Nair

Andrew Yu <andrew at asiatel.com.sg> wrote: it's possible to change the audio path & codec type by sending an 
reINVITE with SDP. could you paste here of the sip trace that you're 
talking about? when an RTP is in session, there is no way that you can 
change the codec type without an reINVITE. G.723.1 5.3kbps and 6.3 kps 
is not inter-compatible and I believe that the SDP should have indicated 
this.


sreekant nair wrote:
> Thanks Boris,
>
> However, the situation is a little bit more tricky.
>
> It was my understanding that codec negotiation is done using the SDP. 
> I tried capturing the INVITE - 18X - 2XX msg using Ethereal to see if 
> there is anything that specifies the bit rate that will be used. In 
> all cases, the codec alone is specified. Even though G.723 supports 
> dual bit rates, I could not find anything which explicitly states the 
> bit rate. (At least Ethereal does not decode it so). Is there 
> something I am missing here.
>
> During a test scenario, both nodes negotiated and agreed on G.723 but 
> one node sent using 5.3kbps while the other replied with 6.3Kbps and 
> the audio path was established. This was found by the amount of data 
> bytes in the RTP packets - one way it was 20 while it was 24 in the 
> reverse direction.
> My question was - Is it possible (both from a hardware/software 
> perspective) to change the bit rates while an RTP session is in 
> progress. So to change the bit rates, there is no need of codec 
> re-negotiation and no UPDATE / RE-INVITE would be sent. Hence my 
> confusion and requirement.
>
> Not sure if I made myself clear.
>
> Sreekant
>
> ----- Original Message ----
> From: Boris vercher 
> To: sreekant nair ; discussion at sipforum.org
> Sent: Friday, July 27, 2007 10:00:56 AM
> Subject: RE: [SIPForum-discussion] G 723 Audio Codec Bit Rate
>
> No it’s not possible , if there are no codec renegotiation
>
>  
>
> There are no compatibility between this two compressions
>
>  
>
> Vercher Boris
>
>  
>
> ------------------------------------------------------------------------
>
> *De :* discussion-bounces at sipforum.org 
> [mailto:discussion-bounces at sipforum.org] *De la part de* sreekant nair
> *Envoyé :* vendredi 27 juillet 2007 15:17
> *À :* discussion at sipforum.org
> *Objet :* [SIPForum-discussion] G 723 Audio Codec Bit Rate
>
>  
>
> Hi All,
>
>  
>
> In our system being tested, one of the codecs used is G.723 audio codec
>
> G.723 has two bit rates - 5.3Kbps & 6.3 Kbps.
>
>  
>
> Is is possible that during a voice call, the codec can dynamically 
> change the bit rate between 5.3Kbps or 6.3 Kbps ?
>
>  
>
> Any thoughts are greatly appreciated.
>
>  
>
> Thanks
>
> Sreekant Nair
>
>  
>
>  
>
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-- 
Cheers,

Asiatel Singapore Pte Ltd
Andrew Yu

19 Jalan Kilang Barat
#06-01, Acetech Centre
Singapore 159361

Tel: +65 6271 8233
Fax: +65 6274 4266



       
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